Takashi Iwai
d47a474aa6
- Remove webrtc-aarch64.patch, no longer needed - Adapt the rest of webrtc- patches to new arch naming - Remove unneeded explicit version dependency for automake - Update to 0.3 * build: enforce linking with --no-undefined, add explicit -lpthread * build: Make sure files with SSE2 code are compiled with -msse2 - Remove no-undefined.patch - Remove webrtc-audio-processing-0.2-x86_msse2.patch - Add no-undefined.patch patch https://cgit.freedesktop.org/pulseaudio/webrtc-audio-processing/patch/?id=d58164e4d87854233564b59e76259b72e21507f6 - Add big_endian_support_2.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738 - Adapt webrtc-audio-processing-0.2-x86_msse2.patch to new version - Adapt big_endian_support.patch to new version - Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-May/026294.html - Add big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738 - New automake version dependency >= 1.5 - Update to 0.2: Contains API breaking changes. Upstream changes include: * Rewritten AGC and voice activity detection * Intelligibility enhancer * Extended AEC filter * Beamformer * Transient suppressor * ARM, NEON and MIPS optimisations (MIPS optimisations are not hooked up) API changes: * We no longer include a top-level audio_processing.h. The webrtc tree format is used, so use webrtc/modules/audio_processing/include/audio_processing.h * The top-level module_common_types.h has also been moved to webrtc/modules/interface/module_common_types.h * C++11 support is now required while compiling client code * AudioProcessing::Create() does not take any arguments any more * AudioProcessing::Destroy() is gone, use standard C++ "delete" instead * Stream parameters are now configured via StreamConfig and ProcessingConfig rather than set_sample_rate(), set_num_channels(), etc. * AudioFrame field names have changed * Use config API for newer audio processing options * Use ProcessReverseStream() instead of AnalyzeReverseStream(), particularly when using the intelligibility enhancer * GainControl::set_analog_level_limits() is broken. The AGC implementation hard codes 0-255 as the volume range Other notes: * The new audio processing parameters are not all tested, and a few are not enabled upstream (in Chromium) either * The rewritten AGC appears to be less sensitive, and it might make sense to initialise the capture volume to something reasonable (33% or 50%, for example) to make sure there is sufficient energy in the stream to trigger the AGC mechanism - Adapted all 3 arch patches OBS-URL: https://build.opensuse.org/request/show/404777 OBS-URL: https://build.opensuse.org/package/show/multimedia:libs/webrtc-audio-processing?expand=0&rev=11
25 lines
905 B
Diff
25 lines
905 B
Diff
diff -up webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef webrtc-audio-processing-0.2/webrtc/typedefs.h
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--- webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef 2016-05-12 09:08:53.885000410 -0500
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+++ webrtc-audio-processing-0.2/webrtc/typedefs.h 2016-05-12 09:12:38.006851953 -0500
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@@ -48,7 +48,19 @@
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#define WEBRTC_ARCH_32_BITS
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#define WEBRTC_ARCH_LITTLE_ENDIAN
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#else
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-#error Please add support for your architecture in typedefs.h
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+/* instead of failing, use typical unix defines... */
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+#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
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+#define WEBRTC_ARCH_LITTLE_ENDIAN
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+#elif __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
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+#define WEBRTC_ARCH_BIG_ENDIAN
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+#else
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+#error __BYTE_ORDER__ is not defined
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+#endif
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+#if defined(__LP64__)
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+#define WEBRTC_ARCH_64_BITS
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+#else
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+#define WEBRTC_ARCH_32_BITS
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+#endif
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#endif
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#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
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